<div dir="ltr">My Dear,<br><br>You have used 'L03' (alphabet 'l' <i>EL</i>) in dial command instead of '103'.<br><br><br>Shariq<br><br><br><div class="gmail_quote">On Mon, Aug 25, 2008 at 3:26 PM, ims.asuser ims.asuser <span dir="ltr"><<a href="mailto:ims.asuser@gmail.com">ims.asuser@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr">Hi all,<br><br>I have a very weird problem.<br><br>I have 2 users (103 and 105). They are able to register in Asterisk, but they can not call each other.<br>
<br>Hereunder is the outcome:<br><br>openwrt3*CLI><br>
-- Registered SIP '103' at <a href="http://192.168.3.9" target="_blank">192.168.3.9</a> port 6127 expires 3600<br> -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 103<br>openwrt3*CLI><br>
openwrt3*CLI><br>
-- Registered SIP '105' at <a href="http://192.168.3.6" target="_blank">192.168.3.6</a> port 8377 expires 3600<br> -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer 105<br>openwrt3*CLI><br>
openwrt3*CLI><br>
-- Executing Dial("SIP/105-0ead", "SIP/l03") in new stack<br>Jan 1 00:19:26 WARNING[498]: chan_sip.c:1407 create_addr: No such host: l03<br>Jan 1 00:19:26 NOTICE[498]: app_dial.c:764 dial_exec: Unable to create channel<br>
of type 'SIP'<br> == Everyone is busy/congested at this time<br>openwrt3*CLI><br>openwrt3*CLI><br> -- Timeout on SIP/105-0ead<br> == CDR updated on SIP/105-0ead<br> -- Executing Goto("SIP/105-0ead", "#|1") in new stack<br>
-- Goto (default,#,1)<br> -- Executing Playback("SIP/105-0ead", "demo-thanks") in new stack<br>Jan 1 00:19:36 WARNING[498]: file.c:475 ast_openstream: File demo-thanks does n<br>ot exist in any format<br>
Jan 1 00:19:36 WARNING[498]: file.c:787 ast_streamfile: Unable to open demo-tha<br>nks (format ulaw): No such file or directory<br>Jan 1 00:19:36 WARNING[498]: app_playback.c:83 playback_exec: ast_streamfile fa<br>iled on SIP/105-0ead for demo-thanks<br>
-- Executing Hangup("SIP/105-0ead", "") in new stack<br> == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-0ead'<br><br><br>The "show sip registry" command shows that no users are registered. That's really WEIRD!<br>
<br><br>Please see the sip.conf and extension.conf files.<br><br>sip.conf:<br><br>[general]<br>context=default ; Default context for incoming calls<br>;recordhistory=yes ; Record SIP history by default<br>
; (see sip history / sip no history)<br>;realm=mydomain.tld ; Realm for digest authentication<br> ; defaults to "asterisk"<br> ; Realms MUST be globally unique according to RF<br>
; Set this to your host name or domain name<br>port=5060 ; UDP Port to bind to (SIP standard port is 5060<br>bindaddr=x.x.x.x ; IP address to bind to (<a href="http://0.0.0.0" target="_blank">0.0.0.0</a> binds to all)<br>
srvlookup=yes ; Enable DNS SRV lookups on outbound calls<br> ; Note: Asterisk only uses the first host<br> ; in SRV records<br> ; Disabling DNS SRV lookups disables the<br>
; ability to place SIP calls based on domain<br> ; names to some other SIP users on the Internet<br><br>[103] ; <br>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!<br>
type=friend<br>username=103 ; Authorization User dans X-Lite<br>secret=1234<br>callerid="Philippe" <103> ; nom et numéro affichés dans le X-Lite appelé l<br>context=default<br>host=dynamic<br>nat=no ; X-Lite is behind a NAT router<br>
canreinvite=no ; Typically set to NO if behind NAT<br>disallow=all ; désactive tous les codages sauf ceux spécifiés ci-aprè<br>allow=gsm ; GSM consumes far less bandwidth than ulaw<br>
allow=ulaw<br>allow=alaw<br><br>[105] ; <br>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!<br>type=friend<br>username=105 ; Authorization User dans X-Lite<br>secret=1234<br>callerid="Khalid" <105> ; nom et numéro affichés dans le X-Lite appelé lor<br>
context=default<br>host=dynamic<br>nat=no ; X-Lite is behind a NAT router<br>canreinvite=no ; Typically set to NO if behind NAT<br>disallow=all ; désactive tous les codages sauf ceux spécifiés ci-aprè<br>
allow=ulaw<br>allow=alaw<br><br><br>extension.conf:<br><br>[default] ; context par défaut des utilisateurs SIP répertoriés dans sip.c<br><br><br>exten => 103,1,Dial(SIP/l03)<br>exten => 105,1,Dial(SIP/l05)<br>
<br></div>
<br>_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
AstriCon 2008 - September 22 - 25 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net" target="_blank">http://www.astricon.net</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>